mirror of
https://github.com/Ryujinx/Ryujinx.git
synced 2024-12-12 01:32:02 +00:00
189 lines
4.9 KiB
C#
189 lines
4.9 KiB
C#
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using ChocolArm64.Memory;
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using Ryujinx.Audio.Adpcm;
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using System;
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namespace Ryujinx.HLE.OsHle.Services.Aud.AudioRenderer
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{
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class VoiceContext
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{
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private bool Acquired;
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private bool BufferReload;
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private int ResamplerFracPart;
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private int BufferIndex;
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private int Offset;
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public int SampleRate;
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public int ChannelsCount;
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public float Volume;
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public PlayState PlayState;
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public SampleFormat SampleFormat;
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public AdpcmDecoderContext AdpcmCtx;
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public WaveBuffer[] WaveBuffers;
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public VoiceOut OutStatus;
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private int[] Samples;
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public bool Playing => Acquired && PlayState == PlayState.Playing;
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public VoiceContext()
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{
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WaveBuffers = new WaveBuffer[4];
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}
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public void SetAcquireState(bool NewState)
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{
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if (Acquired && !NewState)
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{
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//Release.
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Reset();
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}
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Acquired = NewState;
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}
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private void Reset()
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{
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BufferReload = true;
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BufferIndex = 0;
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Offset = 0;
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OutStatus.PlayedSamplesCount = 0;
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OutStatus.PlayedWaveBuffersCount = 0;
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OutStatus.VoiceDropsCount = 0;
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}
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public int[] GetBufferData(AMemory Memory, int MaxSamples, out int SamplesCount)
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{
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if (!Playing)
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{
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SamplesCount = 0;
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return null;
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}
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if (BufferReload)
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{
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BufferReload = false;
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UpdateBuffer(Memory);
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}
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WaveBuffer Wb = WaveBuffers[BufferIndex];
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int MaxSize = Samples.Length - Offset;
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int Size = MaxSamples * AudioConsts.HostChannelsCount;
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if (Size > MaxSize)
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{
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Size = MaxSize;
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}
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int[] Output = new int[Size];
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Array.Copy(Samples, Offset, Output, 0, Size);
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SamplesCount = Size / AudioConsts.HostChannelsCount;
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OutStatus.PlayedSamplesCount += SamplesCount;
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Offset += Size;
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if (Offset == Samples.Length)
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{
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Offset = 0;
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if (Wb.Looping == 0)
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{
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SetBufferIndex((BufferIndex + 1) & 3);
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}
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OutStatus.PlayedWaveBuffersCount++;
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if (Wb.LastBuffer != 0)
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{
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PlayState = PlayState.Paused;
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}
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}
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return Output;
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}
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private void UpdateBuffer(AMemory Memory)
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{
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//TODO: Implement conversion for formats other
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//than interleaved stereo (2 channels).
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//As of now, it assumes that HostChannelsCount == 2.
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WaveBuffer Wb = WaveBuffers[BufferIndex];
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if (SampleFormat == SampleFormat.PcmInt16)
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{
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int SamplesCount = (int)(Wb.Size / (sizeof(short) * ChannelsCount));
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Samples = new int[SamplesCount * AudioConsts.HostChannelsCount];
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if (ChannelsCount == 1)
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{
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for (int Index = 0; Index < SamplesCount; Index++)
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{
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short Sample = Memory.ReadInt16(Wb.Position + Index * 2);
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Samples[Index * 2 + 0] = Sample;
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Samples[Index * 2 + 1] = Sample;
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}
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}
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else
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{
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for (int Index = 0; Index < SamplesCount * 2; Index++)
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{
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Samples[Index] = Memory.ReadInt16(Wb.Position + Index * 2);
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}
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}
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}
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else if (SampleFormat == SampleFormat.Adpcm)
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{
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byte[] Buffer = Memory.ReadBytes(Wb.Position, Wb.Size);
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Samples = AdpcmDecoder.Decode(Buffer, AdpcmCtx);
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}
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else
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{
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throw new InvalidOperationException();
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}
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if (SampleRate != AudioConsts.HostSampleRate)
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{
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//TODO: We should keep the frames being discarded (see the 4 below)
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//on a buffer and include it on the next samples buffer, to allow
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//the resampler to do seamless interpolation between wave buffers.
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int SamplesCount = Samples.Length / AudioConsts.HostChannelsCount;
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SamplesCount = Math.Max(SamplesCount - 4, 0);
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Samples = Resampler.Resample2Ch(
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Samples,
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SampleRate,
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AudioConsts.HostSampleRate,
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SamplesCount,
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ref ResamplerFracPart);
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}
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}
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public void SetBufferIndex(int Index)
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{
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BufferIndex = Index & 3;
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BufferReload = true;
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}
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}
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}
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